When voice data is transported over an IP network, the chosen transport protocol is generally the Real-Time Transport Protocol (RTP). This protocol is conventionally used to transport different kinds of synchronized media, such as video or voice coded with different codecs. RTP is carried over the User Datagram Protocol (UDP). For this reason, the end points of an active voice session using RTP are generally identified using an IP address, i.e. network address, and a UDP port or transport level identifier. The use of RTP enables one or more speech samples to be carried in one RTP package.
The Unlicensed Radio service utilizes an unlicensed radio band to support mobile telecommunication systems operating in licensed radio bands. For example, the Unlicensed Radio service may support Global System for Mobile Communications (GSM) circuit-switched services and GSM Packet Radio Service (GPRS) packet-switched services.
A packet-switched voice session is set-up using a signaling protocol, such as the Session Initiation Protocol (SIP), H-323 or another proprietary or standard protocol. During set-up of the voice session, each party signals to the other party the identifiers that have been locally selected for the voice session.
When a circuit switched call (speech or data) is set up, the network controller specifies the number of speech/data frames that must be used in each RTP package. This is specified as the sample size for the RTP package (which can be converted to a number of 20 ms voice/data samples). If the MS cannot support this requirement, the MS cannot proceed, and the call is terminated.
This problem can be avoided by specifying that a greater number of speech/data frames be used in each RTP package. This enables less capable MSs to proceed with call setup. A disadvantage is that the greater number of speech/data frames in each RTP package increases the delay in the transmission between the MS and the network controller. Thus, the system must be set up either for low delay (some MSs cannot be served) or for higher delay (high performance MSs unnecessarily have higher round trip delay).
Additionally, once a session has begun, there are no procedures for changing the sample size during the ongoing session in order to adapt to changing network conditions.
It would be advantageous to have a system and method that overcomes the disadvantages of the existing methodology. The present invention provides such a system and method.